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libgstopus.so
ELF > R @ P @ 8 @ : : @ @ @ Ł Ł 3 3 P P P $ $ Std Ptd @ @ @ | | Qtd Rtd GNU GNU '9[rH^M MI+Ї m P : S Z { 3 p , m I o / F 6 , + 4 , F " X d M Z $ " \ r 9 I " i : e q | @ # < I ! F A > V i P Z k ) S z U ^ l m r P ` X __gmon_start__ _ITM_deregisterTMCloneTable _ITM_registerTMCloneTable __cxa_finalize _gst_debug_min gst_debug_log gst_debug_category_get_threshold g_string_new g_string_append_printf g_string_append g_string_free g_type_class_peek_parent _gst_debug_register_funcptr gst_element_class_add_static_pad_template gst_element_class_set_static_metadata g_param_spec_boolean g_object_class_install_property _gst_structure_type g_param_spec_boxed g_type_class_adjust_private_offset _gst_debug_category_new g_intern_static_string gst_audio_decoder_get_type g_type_register_static_simple g_once_init_enter g_once_init_leave g_type_check_instance_cast g_type_name g_value_get_boolean opus_multistream_decoder_destroy gst_mini_object_replace gst_audio_decoder_set_plc_aware g_mutex_lock g_mutex_unlock gst_audio_decoder_set_latency gst_audio_decoder_set_needs_format gst_audio_decoder_set_use_default_pad_acceptcaps gst_pad_get_allowed_caps gst_caps_new_empty_simple gst_caps_set_simple gst_caps_intersect gst_mini_object_unref gst_caps_is_empty gst_caps_new_simple gst_caps_truncate gst_caps_get_structure gst_structure_fixate_field_nearest_int gst_structure_get_int memcpy gst_audio_channel_positions_to_valid_order gst_audio_info_init gst_audio_info_set_format gst_audio_decoder_set_output_format _gst_int_range_type __stack_chk_fail gst_buffer_get_size _gst_element_error_printf gst_stream_error_quark gst_element_get_type gst_element_message_full gst_buffer_memcmp gst_buffer_map gst_buffer_unmap gst_codec_utils_opus_parse_header pow gst_pad_get_current_caps gst_caps_is_equal gst_structure_get_value _gst_value_array_type g_type_check_value_holds gst_value_array_get_size gst_codec_utils_opus_parse_caps gst_value_array_get_value _gst_buffer_type g_value_get_boxed g_assertion_message_expr gst_audio_decoder_allocate_output_buffer opus_multistream_decode gst_buffer_set_size gst_audio_clipping_meta_api_get_type gst_buffer_get_meta gst_audio_decoder_finish_frame gst_flow_get_name opus_multistream_decoder_create opus_multistream_decoder_ctl opus_strerror gst_util_uint64_scale_int gst_audio_buffer_reorder_channels gst_buffer_resize _gst_audio_decoder_error gst_library_error_quark g_value_set_boolean gst_structure_new g_value_take_boxed gst_caps_get_size g_value_init gst_value_set_int_range gst_structure_set_value g_value_unset _gst_value_list_type g_value_set_int gst_value_list_append_value gst_mini_object_copy gst_audio_decoder_proxy_getcaps gst_mini_object_make_writable g_enum_register_static g_param_spec_enum g_param_spec_int g_type_check_class_cast g_param_spec_uint gst_type_mark_as_plugin_api gst_audio_encoder_get_type gst_tag_setter_get_type g_type_add_interface_static gst_preset_get_type gst_element_register gst_tag_register_musicbrainz_tags g_value_set_uint g_value_set_enum opus_multistream_encoder_destroy gst_tag_setter_reset_tags gst_event_type_get_name gst_tag_setter_get_tag_merge_mode gst_event_parse_tag gst_tag_setter_merge_tags g_mutex_clear g_mutex_init gst_util_uint64_scale gst_audio_encoder_set_latency gst_audio_encoder_set_frame_samples_min gst_audio_encoder_set_frame_samples_max gst_audio_encoder_set_frame_max g_value_get_uint g_value_get_enum g_value_get_int opus_multistream_encoder_ctl opus_multistream_encoder_create gst_codec_utils_opus_create_header gst_tag_setter_get_tag_list gst_tag_list_to_vorbiscomment_buffer gst_codec_utils_opus_create_caps_from_header gst_audio_encoder_set_output_format gst_tag_list_new_empty gst_tag_list_add gst_audio_encoder_merge_tags gst_audio_encoder_proxy_getcaps gst_caps_new_empty gst_structure_copy gst_structure_set gst_caps_append_structure _gst_bitmask_type gst_structure_free g_malloc0 gst_audio_encoder_allocate_output_buffer opus_multistream_encode gst_audio_encoder_finish_frame gst_buffer_add_audio_clipping_meta gst_plugin_opus_get_desc gst_plugin_opus_register gst_plugin_register_static libgstpbutils-1.0.so.0 libgstaudio-1.0.so.0 libgsttag-1.0.so.0 libgstreamer-1.0.so.0 libgobject-2.0.so.0 libglib-2.0.so.0 libm.so.6 libopus.so.0 libc.so.6 libgstopus.so GLIBC_2.29 GLIBC_2.14 GLIBC_2.4 GLIBC_2.2.5 $ORIGIN/../../gst-libs/gst/pbutils:$ORIGIN/../../gst-libs/gst/video:$ORIGIN/../../gst-libs/gst/audio:$ORIGIN/../../gst-libs/gst/tag ii ui S `S ( 0 8 @ H P X ` h + p A x K V q u y ( 8 @ h p ( 0 @ H X " ` A p , x < J T ] b l x P ( g [ @ ` X d [ ` ` # 3 = B E Q d q ( 0 8 @ H P X ` h p x ! 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HV @ HH+ L PH+ L6 PH H PH H P=H( HH ../ext/opus/gstopusenc.c Unsupported frame size: %d [ %d ] ../ext/opus/gstopuscommon.c %s: %s gst_opus_dec_start gst_opus_dec_stop gst_opus_dec_handle_frame gst_opus_dec_set_format gst_opus_dec_getcaps decode opus streams to audio Codec/Decoder/Audio/Converter Opus audio decoder Use in-band FEC use-inband-fec Apply gain apply-gain Control Phase Inversion phase-inversion Various statistics Statistics opus decoding element opusdec GstOpusDec property ../ext/opus/gstopusdec.c audio/x-raw Negotiated %d channels, %d Hz Using a default of 2 channels OpusHead error: %s warning: %s Channel mapping family %d caps didn't change set_format: %pA Encoding mapping table Decoding mapping table Mono, trivial RTP mapping Stereo, trivial RTP mapping Unknown mapping Mapping table Failed to create Opus decoder Decoding error (%d): %s flow: %s Applying gain: volume %f decoded %d samples FEC disabled, decoding buffer Using buffer of size %lu Using NULL buffer Received buffer of size %lu Received missing buffer %d streams, %d stereo OpusTags found streamheader found vorbiscomments counted vorbiscomments num-gap num-pushed application/x-opusdec-stats last-packet-duration sample-rate bandwidth plc-duration plc-num-samples Unknown bandwith enum: %d Could not retrieve gain: %s GstOpusEncBandwidth GstOpusEncFrameSize GstOpusEncAudioType GstOpusEncBitrateType Encodes audio in Opus format Codec/Encoder/Audio Opus audio encoder gst_opus_enc_start gst_opus_enc_stop gst_opus_enc_set_format gst_opus_enc_handle_frame gst_opus_enc_sink_event gst_opus_enc_sink_getcaps audio-type Encoding Bit-rate bitrate Audio Band Width Frame Size frame-size Bitrate type bitrate-type Complexity complexity In-band FEC DTX dtx Packet loss percentage Loss percentage packet-loss-percentage Maximum payload size in bytes Max payload size max-payload-size gst_opus_enc_finalize Opus encoder opusenc GstOpusEnc sink event: %s init Latency: %u:%02u:%02u.%09u Encoder creation failed here are the caps: %pA Setup with %d channels, %d Hz layout S16LE interleaved channel-mask Returning caps: %pA sink getcaps Encoding failed (%d): %s Output packet is %u bytes size == bytes bsize < bytes draining %d samples nothing to drain 1.22.12 OPUS plugin library gst-plugins-base LGPL mono front left front right rear center rear left rear right lfe front center front left of center front right of center side left side right none sink src audio/x-opus CBR cbr Constrained VBR constrained-vbr Generic audio generic Voice voice Restricted low delay restricted-lowdelay 2.5 10 20 40 60 Narrow band narrowband Medium band mediumband Wide band Super wide band superwideband Full band fullband Auto auto 2024-04-29 yyyyy zyyyy zyyyyyyyyy@zyyyyyyyyyyyyyyyyyyy`z ,<Tlp`P@ gst_opus_enc_encode gst_opus_enc_handle_frame gst_opus_enc_sink_getcaps gst_opus_enc_sink_event gst_opus_enc_setup gst_opus_enc_set_format gst_opus_enc_setup_channel_mappings gst_opus_enc_find_channel_position_in_vorbis_order gst_opus_enc_get_frame_samples gst_opus_enc_get_latency gst_opus_enc_stop gst_opus_enc_start gst_opus_enc_init get_last_packet_duration get_gain get_sample_rate get_bandwidth gst_opus_dec_handle_frame gst_opus_dec_set_format packet_duration_opus @ @ @ x x x x opus_dec_chain_parse_data gst_opus_dec_parse_header gst_opus_dec_negotiate gst_opus_common_log_channel_mapping_table Channel position %s maps to index %d in Vorbis order Channel position %d is not representable in Vorbis order Vincent Penquerc'h <vincent.penquerch@collabora.co.uk> Use forward error correction if available (needs PLC enabled) Apply gain if any is specified in the header Set to true to enable phase inversion, this will slightly improve stereo quality, but will have side effects when downmixed to mono. %s:%d: invalid %s id %u for "%s" of type '%s' in '%s' Empty intersection, failed to negotiate Using a default of 48kHz sample rate Header is not an Opus ID header Failed to parse Opus ID header Using NONE channel layout for more than 8 channels Channel mapping family 1, Vorbis mapping Found pre-skip of %u samples, R128 gain %d (volume %f) caps have changed, resetting decoder Cannot map channel positions to Vorbis order, using unknown mapping Natural pair '%s/%s' found at %d %d Channel position %d is not mapped yet, adding Mapping tables built: %d channels, %d stereo streams In range for the Vorbis mapping, building channel mapping tables Unpositioned mapping, all channels mono Setting up channel mapping for %d channels Failed to create Opus decoder (%d): %s missing buffer, doing PLC duration %u:%02u:%02u.%09u plus leftover %u:%02u:%02u.%09u current duration %u:%02u:%02u.%09u of missing data not enough for PLC (minimum needed: %u:%02u:%02u.%09u) - skipping calculated PLC frame length: %u:%02u:%02u.%09u num frame samples: %d new leftover: %u:%02u:%02u.%09u Failed to create %u byte buffer First buffer received in FEC mode, early out Skipping %u samples at the end (%u at 48000 Hz) Skipping %u samples at the beginning (%u at 48000 Hz) !cmeta || cmeta->format == GST_FORMAT_DEFAULT FEC enabled, decoding last delayed buffer FEC enabled, reconstructing last buffer Code 3 Opus packet has less than 2 bytes Opus packet duration > 120 ms, invalid Opus packet: frame size %.1f ms, %d frames, duration %.1f ms Missing duration, using last duration %u:%02u:%02u.%09u Missing buffer, but unknown duration, and no previously known duration, assuming 20 ms Could not configure phase inversion: %s Creating decoder with %d channels, %d Hz No header, assuming single stream Got buffer ts %u:%02u:%02u.%09u, duration %u:%02u:%02u.%09u Could not retrieve bandwith: %s Could not retrieve sample rate: %s Could not retrieve last packet duration: %s What type of audio to optimize for Specify an encoding bit-rate (in bps). The duration of an audio frame, in ms Enable in-band forward error correction (use in combination with the packet-loss-percentage property) Encoded with GStreamer opusenc we have frame size %d, lookahead %d setup: %d Hz, %d channels, %d stereo streams, family %d received buffer %p of %lu bytes Encoded size %d is higher than max payload size (%d bytes) Adding trim-start %lu trim-end %lu encoding %d samples (%d bytes) %li extra samples of padding in this frame Need to add %li extra samples in the next frame draining; adding silence samples http://download.fedoraproject.org Fedora GStreamer-plugins-base package audio/x-raw, format = (string) S16LE, layout = (string) interleaved, rate = (int) 48000, channels = (int) [ 1, 8 ]; audio/x-raw, format = (string) S16LE, layout = (string) interleaved, rate = (int) { 8000, 12000, 16000, 24000 }, channels = (int) [ 1, 8 ] audio/x-opus, channel-mapping-family = (int) 0; audio/x-opus, channel-mapping-family = (int) [1, 255], channels = (int) [1, 255], stream-count = (int) [1, 255], coupled-count = (int) [0, 255] audio/x-raw, format = (string) S16LE, layout = (string) interleaved, rate = (int) { 48000, 24000, 16000, 12000, 8000 }, channels = (int) [ 1, 8 ] p? 4@ $@ ? @B $tI ;| . 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